[asterisk-users] asterisk-users Digest, Vol 125, Issue 33
viljoens at verishare.co.za
Tue Dec 30 00:22:51 CST 2014
(please excuse me for lack of proper jargon usage and the vagueness of
i use Asterisk 11.12.1, (well... as included in FreePBX),
The softphones are mostly on machines without proper sound hardware (no
mics, no speakers/headsets); This is partly because the workforce is quite
conservative in what they want to use :) meaning handsets are important;
As the handsets have no LCD's to show the dialled number, I want to give the
workforce the ability to dial OUT using the softphone, (as in, copy/paste
the number from the CRM software into softphone then
*immediately* transfer the originated call 'endpoint' to the handset of the
same 'user' extension, somehow, the question is, HOW ?
I think you're overcomplicating your problem. (if I understand you
Your scenario is almost exactly ours, except we use ATCOM-820P's (with LCD
displays) and no softphones. So incoming CID is displayed on the phone's
physical LCD displays.
What we did is write our own C# dialler app - all this effectively does
(through a third-party server app we designed) is connect over the AMI to
the Asterisk instance and then use the "originate" function to originate a
call to the user's phone.
Behind this is a database where we store which logged in user in the dialler
app is which extension - e. g. by updating the DB we can "send" a call
originated by one user "anywhere" among the group of SIP phones connected to
E. g. I think you can do this too?
Instead of them copying the number into the softphone (causing all your SIP
pain / confusion to get the "real" phone to then ring with an outgoing call
queued to that number) have a second app running (it can be TINY - both in
amount of code and on-screen presence) - that does an AMI originate with the
Asterisk and sends the desktop originated call to the relevant hardphone?
Thereby avoiding the extremely complicated SIP setup / manipulation you want
Just a thought.
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